| Friday March 31 2006 |
| Time | Replies | Subject |
|---|
| 11:52PM |
0 |
asterisk-stat and webmeetme by areski |
| 9:40PM |
0 |
testing list mail - please ignore |
| 6:36PM |
1 |
Zap channels - help |
| 4:56PM |
0 |
Re: BRI cards, HFC, and bristuff - a general questionto clear up my understanding. |
| 4:49PM |
0 |
Re: Asterisk-Users Digest, Vol 20, Issue 226 |
| 3:21PM |
2 |
Iaxmodem speed limit? |
| 1:55PM |
8 |
1.2.6 doesn't use mpg123? |
| 1:17PM |
3 |
Display Name |
| 1:07PM |
1 |
Play wav while in connection with a caller |
| 12:54PM |
0 |
OT: ad-hoc polycom network |
| 12:08PM |
0 |
hosted billing service |
| 11:47AM |
0 |
MeetAsterisk Europe:: Get an Asterisk one-day introduction! |
| 11:03AM |
0 |
chan_nbs and nbs |
| 10:57AM |
5 |
Dial from php |
| 10:53AM |
0 |
Transcoding on asterisk |
| 10:42AM |
1 |
incoming triggers seperate outbound |
| 10:29AM |
2 |
Asterisk Referral - Cleanup on Aisle 7 |
| 10:24AM |
1 |
Confused on Agents and Queues |
| 9:33AM |
2 |
PRI issues |
| 9:06AM |
3 |
Echo cancellation problem |
| 8:58AM |
1 |
transcoding g723 or g729 on asterisk |
| 8:56AM |
1 |
statechange_queue |
| 8:11AM |
0 |
decrease the speed of reading text!!! |
| 8:08AM |
4 |
cannot set outgoing cid |
| 7:57AM |
2 |
IAXY codec support and questions.. |
| 7:55AM |
0 |
meetme option 'e' |
| 7:47AM |
1 |
Asterisk hosted solution |
| 7:15AM |
1 |
oh323 - unable to install |
| 7:01AM |
10 |
Building Asterisk embedded device |
| 7:00AM |
1 |
I have debug off why are the logs show debug info |
| 6:05AM |
1 |
Echo still present with Eicon Diva Server 4 Bri |
| 5:53AM |
4 |
IAX: Auto-congesting call due to slow response |
| 5:16AM |
1 |
Re: BRI cards, HFC, and bristuff - a generalquestionto clear up my understanding. |
| 5:03AM |
3 |
Howto cut the first digit |
| 4:30AM |
1 |
Re: BRI cards, HFC, and bristuff - a general questionto clear up my understanding. |
| 4:24AM |
0 |
How do you perform a Variable Substitution In Asterisk |
| 4:01AM |
1 |
Asterisk, QSIG and Tenovis PBX? |
| 3:48AM |
3 |
Quintum Tenor DX4060 |
| 3:24AM |
1 |
Outgoing SIP Failover |
| 3:07AM |
0 |
No voice heard in festivalassociated with asterisk!!! |
| 2:41AM |
3 |
asterisk turn key solution |
| 2:29AM |
0 |
bristuff does not work with TDM400P |
| 1:09AM |
4 |
How to check if a phone / line is used? |
| |
| Thursday March 30 2006 |
| Time | Replies | Subject |
|---|
| 11:38PM |
1 |
Multicast Music on Hold |
| 9:46PM |
1 |
Using Voicemail with MP3 files... |
| 9:17PM |
1 |
DID billing |
| 8:16PM |
3 |
Anybody know about Cisco VOIP routers? |
| 7:08PM |
1 |
caller anounce |
| 5:24PM |
1 |
Questions on call recording and conference. |
| 4:49PM |
0 |
SIP Plugin or Component |
| 3:54PM |
3 |
Config TE110P and TDM400 with 2 FXS modules |
| 3:07PM |
1 |
Zaptel compile errors |
| 3:03PM |
2 |
multiple auto attendants |
| 2:39PM |
5 |
Asterisk and Hylafax, on the same box |
| 2:33PM |
1 |
OT: Polycom IP501 and Speed Dials |
| 2:23PM |
0 |
Vmail.cgi and #include |
| 2:19PM |
1 |
misdn timeout? |
| 2:13PM |
3 |
Please Help Test Quad PRI Using NFAS |
| 1:28PM |
1 |
Authenticate |
| 1:22PM |
0 |
PRI channel hangs after BUSY |
| 1:11PM |
0 |
DID's Now Offering Romania Bucharest 4021+ and 4031+ |
| 1:08PM |
0 |
How we tell who is using VAD ? |
| 1:05PM |
1 |
Disable polycom call waiting? |
| 1:02PM |
0 |
BUG: FOP reports incorrect (duplicate) IP address until restarted |
| 12:38PM |
5 |
Reload astdb? |
| 11:13AM |
1 |
'sip show users' shows NAT RFC3581 |
| 11:11AM |
1 |
sending SIP text messages to capable phones from an app |
| 10:28AM |
0 |
AMP backup-restore problem |
| 9:58AM |
1 |
internals and ISDN calls fail when Internet is down |
| 9:34AM |
0 |
Wrong extension indicated when logging in as agent |
| 9:28AM |
1 |
Benchmarking an Asterisk Server with 14k users |
| 9:24AM |
1 |
Sjphone looses registry in my Asterisk Server then i need to restart Sjphone PC, why? |
| 9:15AM |
0 |
SIP: INFO before answer causes disconnect |
| 9:14AM |
9 |
How is Teliax ? |
| 9:02AM |
2 |
Connecting a Grandstream Handytone 486 to Asterisk |
| 8:56AM |
0 |
Setting up announcement on reply to 4xx 5xx 6xx messages |
| 8:42AM |
3 |
Span monitoring |
| 7:46AM |
0 |
Strange second REINVITE being sent |
| 7:20AM |
4 |
Asterisk in production as a fax server, anyone? |
| 7:01AM |
3 |
asterisk doesn't wait for whole extension |
| 6:18AM |
3 |
Callid on T-1 trunk |
| 4:07AM |
1 |
Asterisk out of Media Path - Call Park |
| 3:32AM |
2 |
TDM04B sound volume |
| 3:27AM |
1 |
Panasonic KXTD 1232 6 |
| 1:43AM |
0 |
Why would asterisk presume a loop (482 "Loop Detected")? |
| 12:53AM |
0 |
mixmonitor and how to stop it |
| 12:45AM |
0 |
Any new Voice Recognition devs? |
| 12:28AM |
0 |
Re: Asterisk-Users Digest, Vol 20, Issue 211 |
| |
| Wednesday March 29 2006 |
| Time | Replies | Subject |
|---|
| 11:58PM |
0 |
R: RE : Echo cancellation |
| 11:56PM |
1 |
SV: IAX - only one way traffic |
| 10:40PM |
5 |
Problem with setting ringtones on Cisco 7960 phone. |
| 10:01PM |
1 |
Realtime Users/Peers/Friends - Ick |
| 8:01PM |
0 |
src/chan_h323.c: Failed to initialize OOH323 |
| 6:11PM |
1 |
Blacklist out bound numbers from file |
| 4:52PM |
1 |
OT: FOP and reverse_transfer |
| 4:52PM |
2 |
Asterisk as Voicemail Server for Option 61c? |
| 4:34PM |
0 |
eyeBeam v1.1 |
| 3:38PM |
5 |
Asterisk Between PBX and FXS |
| 3:37PM |
4 |
Dumb question - reaching the PSTN |
| 2:02PM |
0 |
Streaming voice using IAX |
| 1:51PM |
0 |
Two X100p clones. One not available for outbound? |
| 12:58PM |
1 |
Unable to open Asterisk database |
| 12:19PM |
4 |
Regulatory Ruling about Caller-ID |
| 12:12PM |
1 |
Inter-Asterisk Using SIP |
| 11:29AM |
2 |
H323 behind a Firewall |
| 10:59AM |
2 |
Calling home while on the road, will it work? |
| 10:35AM |
1 |
zaphfc on an 'actual' asterisk? |
| 10:08AM |
1 |
OT: HOWTO: Query channel state on an Ateus Voice Blue GSM gateway |
| 9:42AM |
0 |
indications.conf.sample |
| 9:34AM |
1 |
SJphone Do not send silence - option ? Should be disabled for Asterisk |
| 9:32AM |
0 |
Installing Cisco IP phone 7910 |
| 9:26AM |
1 |
cdr_odbc appears to have fields missing |
| 9:02AM |
2 |
AAH lost my IVR phrases |
| 9:01AM |
3 |
FOP flash panel: how to reload config files when running |
| 8:45AM |
2 |
IAX - only one way traffic |
| 8:35AM |
0 |
inbound routing help |
| 8:08AM |
6 |
Routing SIP calls via URI |
| 7:37AM |
4 |
Marketing Materials |
| 6:59AM |
6 |
Asterisk with Vonage |
| 6:55AM |
3 |
SMS in Spain (it seems Protocol 2) |
| 6:51AM |
1 |
Oneway Audio |
| 6:05AM |
7 |
Reporting? |
| 2:06AM |
1 |
Avoiding initial deadlock on iax? |
| |
| Tuesday March 28 2006 |
| Time | Replies | Subject |
|---|
| 11:12PM |
0 |
Realtime mapping problem after svn upgrade |
| 7:59PM |
3 |
Call Monitoring / Call Takeover with Asterisk |
| 6:21PM |
0 |
IAX2 errors |
| 6:05PM |
3 |
dial plan logic |
| 5:22PM |
1 |
Dialogic d/4 PCI |
| 4:12PM |
0 |
H323 Info |
| 4:04PM |
1 |
Asterisk Tools for OSX |
| 4:03PM |
2 |
Transferring calls - BUG0003710 |
| 3:58PM |
4 |
RTP frame size location? |
| 3:31PM |
0 |
Can realtime extensions be used within AEL contexts? |
| 3:22PM |
1 |
Asterisk to MySQL Data Lookup Warning Message? |
| 2:47PM |
1 |
RXgain |
| 1:12PM |
2 |
Problems Configuring Cisco 12SP+ |
| 1:08PM |
2 |
Asterisk & SMP: Is irqbalance Redundant on 2.6 Kernels? |
| 1:05PM |
1 |
Redirect problem/bug/feature |
| 1:05PM |
1 |
AAH Mailing list |
| 12:57PM |
3 |
How to send announcement after called has picked up the phone? |
| 12:26PM |
3 |
Softphone accepting URL |
| 12:07PM |
1 |
Asterisk eating CPU |
| 11:48AM |
0 |
Bluetooth headsetin handsfree modewith SJPhoneorX-lite |
| 10:40AM |
2 |
Asterisk 1.2.6, VMWare, & Playback/Background GSM prompts |
| 10:39AM |
0 |
WARNINGS For SIP call |
| 10:33AM |
1 |
Monitoring question |
| 10:21AM |
0 |
R: R: Echo cancellation |
| 10:05AM |
1 |
Psgw |
| 10:01AM |
0 |
CID passthrough |
| 9:44AM |
3 |
Unable to authenticate password - VM |
| 8:41AM |
0 |
Bluetooth stack for cordless telephone |
| 8:29AM |
2 |
NATted phones transferring calls - BUG0003710 |
| 7:53AM |
3 |
R: Echo cancellation |
| 7:51AM |
2 |
Agents on DND still receiving calls... |
| 7:45AM |
3 |
aah 2.7 / BRI |
| 7:32AM |
1 |
Asterisk 1.2.6 on Solaris 8? |
| 7:27AM |
0 |
h323 channel driver for production |
| 7:25AM |
0 |
codec translation problem??? |
| 7:13AM |
3 |
Agent in multiple queues? |
| 5:28AM |
1 |
Squished faxes with txfax |
| 5:19AM |
3 |
Set caller ID for outgoing PRI calls |
| 5:15AM |
0 |
Injecting faxes directly to queue? |
| 4:49AM |
0 |
DTMF recognition inconsistent in Asterisk |
| 4:48AM |
1 |
IAX problems - please help me |
| 4:31AM |
4 |
Problems with wcte11xp module |
| 3:05AM |
0 |
Addons 1.2.1 upgrade to 1.2.2 |
| 2:47AM |
2 |
Dial out .call files File permissions?? |
| 2:38AM |
4 |
ISDN and Analog DIAL UP Connection Through Asterisk and Digium TE405P |
| 12:16AM |
1 |
time update (7905) |
| |
| Monday March 27 2006 |
| Time | Replies | Subject |
|---|
| 11:35PM |
0 |
how to dial to hardphone by kiax |
| 8:39PM |
0 |
TDM11B desperate Help wanted |
| 8:11PM |
1 |
restart problem |
| 5:38PM |
0 |
BUG 0003710 - RE: Transfer Calls - REFER |
| 5:19PM |
3 |
Dell 2850 w/TDM2400? |
| 4:42PM |
2 |
definity g3 voicemail |
| 4:40PM |
0 |
Transfer Calls - REFER |
| 4:27PM |
0 |
SER inside of Asterisk is SCALABLE ? |
| 4:17PM |
1 |
Bluetooth headset in handsfree modewith SJPhoneorX-lite |
| 3:51PM |
0 |
ANNOUNCE: WIST - Web Interface for SIP Trace |
| 3:45PM |
0 |
oh323 signal update support |
| 3:31PM |
2 |
How to disable event_log? |
| 3:20PM |
3 |
sipura spa2 + asterisk bug ? |
| 3:14PM |
2 |
TDM400P busy |
| 2:36PM |
0 |
Intel Compilation Questions - Asterisk 1.2.5/6 |
| 1:29PM |
2 |
FXO without answer supervision |
| 12:59PM |
1 |
Bluetooth headset in handsfree modewith SJPhoneor X-lite |
| 12:55PM |
1 |
queue caveats |
| 12:43PM |
0 |
TE 205P/A102 fit in hp dc7600? |
| 12:25PM |
1 |
Master.csv Shell Script |
| 12:18PM |
2 |
Call Waiting Issues |
| 12:09PM |
0 |
Wanted: Cd-bootable Fedora+Asterisk |
| 12:06PM |
5 |
Alarm on Unicall |
| 11:51AM |
1 |
Asterisk 1.2.6 and Zaptel 1.2.5 Released |
| 11:32AM |
2 |
Receptionist Phones (was 3Com Phones) |
| 11:23AM |
1 |
Searchable forums |
| 11:20AM |
2 |
Ability to put call on hold via manager? |
| 10:46AM |
0 |
Unicall Question |
| 10:15AM |
5 |
FreePBX & AAH |
| 9:55AM |
0 |
SIP caller id |
| 9:48AM |
1 |
Polycoms and hints |
| 9:36AM |
3 |
CLI Echo |
| 9:01AM |
2 |
Testing asterisk faxing functionality |
| 8:57AM |
0 |
Inaudible voice and sleepy voice |
| 8:48AM |
3 |
Config File Management |
| 8:37AM |
1 |
after-queues |
| 8:19AM |
1 |
Small - Medium Billing Software needed |
| 8:04AM |
1 |
Bluetooth headset in handsfree mode with SJPhoneor X-lite |
| 7:21AM |
0 |
Bluetooth headset in handsfree mode with SJPhone or X-lite |
| 7:05AM |
1 |
FW: Re: Fw: anybody has SIP realtime working ? |
| 6:41AM |
0 |
Timeout waiting for response to Originate |
| 5:46AM |
1 |
automatic callback when busy |
| 5:27AM |
0 |
Who hangup. |
| 5:21AM |
0 |
Question about Polycom 601 and expansion module. |
| 4:56AM |
2 |
Voicemail to Email |
| 4:51AM |
0 |
Re: Re: Cisco 7960 - Have to press a menu button to dial |
| 4:47AM |
0 |
Transfer after group pick-up |
| 4:38AM |
2 |
Call Simulator |
| 4:24AM |
2 |
registration with different username |
| 3:57AM |
4 |
Polycom 501 Output volume |
| 3:08AM |
1 |
Caller ID length |
| 1:57AM |
2 |
Any Polycom dealer willing to help? |
| 1:49AM |
0 |
get no connection, very often, but not allways, why? |
| 1:35AM |
0 |
iax2_poke_noanswer on IP change. Sometimes permanent. |
| 1:34AM |
4 |
Alarmreciver |
| 12:08AM |
1 |
7940 with Asterisk? |
| |
| Sunday March 26 2006 |
| Time | Replies | Subject |
|---|
| 11:22PM |
1 |
Re: Cisco 7960 - Have to press a menu button to dial |
| 11:22PM |
2 |
Free g729 |
| 11:22PM |
0 |
Asterisk add-ons upgrade |
| 7:06PM |
2 |
Web based voicemail client |
| 5:57PM |
1 |
Snom 360 - Multiple Server BLF Indications |
| 5:41PM |
0 |
RE: Hopefully a Simple Question? |
| 4:56PM |
0 |
Jittery Linksys/Sipura meetme conference fixed |
| 4:07PM |
0 |
UK EI |
| 3:03PM |
0 |
SIP realtime: how to authenticate without "name" field ? |
| 2:57PM |
0 |
MusicOnHold with mpg123 |
| 11:00AM |
0 |
RE: Asterisk-Users Digest, Vol 20, Issue 184 |
| 10:44AM |
1 |
AAH: DNID not set if caller suppresses CID? |
| 10:30AM |
0 |
hang up when pickup analog phone |
| 9:19AM |
2 |
tsu-600 |
| 8:20AM |
0 |
Polarity reversals on a TE100P |
| 6:14AM |
0 |
zapata configuration & parsing |
| 3:28AM |
1 |
What codec extensions using now? |
| |
| Saturday March 25 2006 |
| Time | Replies | Subject |
|---|
| 11:28PM |
2 |
Copying SIP Subscriptions |
| 11:06PM |
1 |
WARNING[5171]: res_musiconhold.c:833 moh_register: Unable to open pseudo channel for timing |
| 5:28PM |
0 |
Mailing list problems with gmail!!!! |
| 5:23PM |
0 |
CLI notice: channel.c:2421 __ast_request_and_dial: Don't know what to do with control frame 15 |
| 3:59PM |
4 |
Asterisk and "Commercial Unix" |
| 3:15PM |
2 |
Disable timeout for answered queue calls? |
| 3:06PM |
2 |
On site installtion Tech. wanted |
| 2:39PM |
5 |
compiling Zaptel-1.2.4 on CentOS 4.3 |
| 7:55AM |
2 |
Asterisk billing from CDR database |
| 5:23AM |
3 |
QuesCom 400 IP/GSM |
| 4:56AM |
2 |
Asterisk spanDSP / Faxing problem |
| 4:45AM |
0 |
WARNING[3675] samples/codec_g723.c: Received a G.723.1 frame that was 4 bytes from RTP |
| 2:45AM |
0 |
apic vs xt-pic on fc3? |
| 2:36AM |
2 |
help on mfc/r2 |
| 1:22AM |
1 |
Error in starting * with latest trunk |
| 1:06AM |
6 |
Polycom IP 301 is slow |
| |
| Friday March 24 2006 |
| Time | Replies | Subject |
|---|
| 10:53PM |
14 |
IAX Incoming/Outgoing |
| 9:24PM |
0 |
Bluetooth headset with SJPhone |
| 7:03PM |
3 |
iax limit question |
| 6:21PM |
5 |
GSM/DECT handsets (was gsm picocells) |
| 6:10PM |
3 |
Polycom 601 Message Center |
| 5:47PM |
1 |
[1.2.5] DTMF not being set correctly (RESEND) |
| 5:19PM |
2 |
3Com Phones |
| 3:28PM |
1 |
Re: Subscription state after reload (New subject) |
| 3:11PM |
2 |
SV: re: Sound issues on SIP-SIP calls |
| 2:51PM |
1 |
PRI Behavior |
| 2:19PM |
2 |
SIP trunk problem |
| 1:17PM |
0 |
Finding the busypattern |
| 12:40PM |
11 |
Transferring a call with IAX |
| 12:01PM |
1 |
making ooh323 authenticate gateway just like sip does |
| 11:42AM |
0 |
FW: Extension a? |
| 11:24AM |
0 |
Queue Period-Announce |
| 11:18AM |
0 |
Realtime Agents |
| 11:11AM |
0 |
VoIP QoS monitoring and failover re-routing |
| 10:55AM |
8 |
Asterisk Failover without SER |
| 10:47AM |
1 |
Maximum Queue Name Length |
| 10:42AM |
1 |
FW: Asterisk Users |
| 10:41AM |
8 |
Snom 360 problems |
| 10:23AM |
0 |
Echo and static when dialing Asterisk |
| 10:03AM |
3 |
Best GUI for basic HostedPBX service |
| 9:10AM |
2 |
Getting True ANI not Caller ID |
| 8:25AM |
1 |
Plain Old Answering Machine |
| 8:05AM |
3 |
* Meetme Freeze patch found |
| 7:43AM |
0 |
On ParkAndAnnounce and parking lot |
| 6:37AM |
5 |
Mandrake zaptel module not found after compiling |
| 6:25AM |
0 |
Hints in Realtime |
| 5:11AM |
0 |
RES: reload - restart |
| 4:49AM |
1 |
reload - restart |
| 4:01AM |
3 |
Call terminated after 60 seconds |
| 3:35AM |
0 |
Which 2 Port ISDN Card for P2P (Austria) |
| 3:00AM |
1 |
Re: Server freeze with meetme and sip GSM users |
| 2:50AM |
2 |
How to nice agi scripts? |
| 2:48AM |
0 |
Speed up dial using #? |
| 2:21AM |
0 |
UK pri almost working |
| 2:07AM |
1 |
Problem with MeetMe Conference!!! |
| 1:27AM |
0 |
pots -> asterisk -> tsu-600 |
| 12:42AM |
1 |
chan_h323 problem |
| 12:36AM |
1 |
getting your own phone number |
| 12:09AM |
1 |
problems compiling zaptel on FC5 |
| |
| Thursday March 23 2006 |
| Time | Replies | Subject |
|---|
| 11:59PM |
0 |
CallerID chopped by half ? :-) |
| 9:57PM |
0 |
What do the Queue timeouts really mean? |
| 9:29PM |
1 |
Page about 70 users crash my Asterisk |
| 9:25PM |
6 |
Stability of Asterisk with 2 x TDM400P cards (6 analogue lines) |
| 8:44PM |
0 |
GnuGk and Asterisk IVR |
| 8:26PM |
0 |
realtime and queues and persistantmembers in 1.2.5 |
| 7:57PM |
1 |
IAX Bridging and not recording CDR correctly |
| 7:17PM |
8 |
FXS channel banks |
| 6:11PM |
3 |
Which g729 codec to download for a P4? |
| 6:00PM |
9 |
Tearing my hair out with Queues |
| 5:39PM |
3 |
Polycom 501's for sale |
| 4:48PM |
4 |
Which Mac OSX softphone with IAX2 support? |
| 3:50PM |
7 |
[OT] Polycom provisioning |
| 3:35PM |
0 |
7940/60 SIP Call Park Button |
| 3:34PM |
0 |
Re: Subscription state after reload (New subject) |
| 3:29PM |
1 |
kernel recompilation on a asterisk server |
| 3:28PM |
0 |
Re: Subscription state after reload (New subject) |
| 3:21PM |
1 |
transfer incoming call to VM without answering call |
| 3:01PM |
0 |
Anonymous sip calls getting into wrong context? |
| 2:57PM |
1 |
wellgate 38XX FX & FXS voip gateways with outgoing call files |
| 2:45PM |
1 |
User Extension Custom Voicemail |
| 2:32PM |
1 |
Dialling Problem |
| 2:25PM |
2 |
Changing codec. |
| 1:42PM |
2 |
TAC Case Cisco 7960 Proxy address showing up in callerID |
| 1:32PM |
3 |
Aastra 9331i phones |
| 12:40PM |
0 |
Cracked and sleepy voice |
| 12:18PM |
6 |
How to create [new_context] in extensions.conf? |
| 12:05PM |
1 |
RE: MeetMe freezes machine with Junghanns |
| 12:05PM |
6 |
I'm FED UP with BroadVoice |
| 11:54AM |
1 |
High Density Analog |
| 11:25AM |
3 |
MeetMe freezes machine with Junghanns QuadBRI cards |
| 11:16AM |
7 |
Ok... what is 'sip show peers' really used for? |
| 10:57AM |
1 |
Call Monitoring? |
| 10:53AM |
3 |
Call Recording? |
| 10:24AM |
1 |
Cisco 7970 SIP Image - hint lines |
| 10:19AM |
1 |
"Not Found" in archive |
| 9:54AM |
0 |
Problem with INVITE's being sent |
| 9:52AM |
0 |
Arcavox / AV-Global |
| 9:51AM |
1 |
Problem with Queue periodic announcemnets |
| 9:00AM |
1 |
SIP - Problem with audio clipping |
| 8:41AM |
0 |
How to monitor dropped frames on a PCI bus |
| 8:38AM |
2 |
Maximum retries exceeded on transmission |
| 8:31AM |
0 |
Billing from CDR files |
| 8:10AM |
0 |
Zaptel compilation problem on SUSE |
| 8:09AM |
0 |
compiling chan_h323 |
| 7:55AM |
0 |
Sending 2 CallingNumbers |
| 7:28AM |
0 |
Trouble installing a digium TE405P card |
| 5:50AM |
3 |
best MTU? |
| 4:34AM |
1 |
CHINA DID |
| 4:05AM |
2 |
asterisk as a fax server |
| 2:29AM |
1 |
Netgear FS116P and Cisco 79XX phones |
| 1:21AM |
2 |
type of incoming lines |
| |
| Wednesday March 22 2006 |
| Time | Replies | Subject |
|---|
| 11:24PM |
2 |
welltech Wellgate 3804 in SIP mode |
| 10:54PM |
0 |
Video phone failed on Asterisk-1.2.4 |
| 9:40PM |
0 |
AstriCon Europe: Early Bird Open / Speakers & Papers Wanted |
| 9:25PM |
2 |
G729 License questions |
| 8:16PM |
1 |
Hardware Levels |
| 7:41PM |
2 |
polycom queue bug |
| 7:21PM |
1 |
Polycom IP501 Buddy List |
| 7:17PM |
3 |
7970 8.x firmware speeddials |
| 5:49PM |
2 |
Re: Asterisk-Users Digest, Vol 20, Issue 153 |
| 4:21PM |
3 |
PRI DMS100 -> Nortel Meridian Option 81 |
| 4:01PM |
2 |
connecting Avaya Partnet with asterisk , TE205P |
| 3:52PM |
3 |
Asterisk Users |
| 2:58PM |
1 |
pseudo Direct Outward Dial |
| 2:50PM |
1 |
Stability and motherboard questions with TE406P and TE410P |
| 2:45PM |
1 |
best CENTOS to use for latest asterisk |
| 12:50PM |
3 |
Voicemail limit? |
| 11:57AM |
6 |
Can this box handle 8 T1s (PSTN) with Asterisk? |
| 11:05AM |
7 |
VERY IMPORTANT(TREAT WITH URGENCY) |
| 10:47AM |
1 |
Big Traffic anyway? |
| 9:55AM |
2 |
Realtime Query |
| 9:39AM |
0 |
call drops after one ring |
| 9:24AM |
3 |
router UDP timeout |
| 9:00AM |
1 |
Dial plan question - exclamtion mark |
| 8:54AM |
1 |
Asterisk snapshots? |
| 8:43AM |
0 |
SPA-2002 Upgrade Question |
| 8:38AM |
2 |
Asterisk--->>Autodialling |
| 8:34AM |
3 |
what are these and can they be fixed? |
| 8:31AM |
1 |
License for asterisk-addons? |
| 8:28AM |
0 |
TIMEOUT(s) |
| 8:10AM |
3 |
Remote dialtone |
| 7:47AM |
0 |
the best configuration for DTMF detection on SPA 2000 |
| 7:18AM |
0 |
RE: VoiceMailMain(@context) Problem with Opt ion 5 (Advanced) |
| 7:07AM |
0 |
ZOMBIE on att transfer |
| 6:31AM |
4 |
Sound issues on SIP-SIP calls |
| 6:27AM |
5 |
Double Call Progress tones |
| 5:04AM |
2 |
beronet & bristuff |
| 4:50AM |
1 |
How to hide CallerID - SetCallerPres(prohib) not working |
| 4:44AM |
1 |
Asterisk & Avaya Legend |
| 3:54AM |
2 |
Asterisk perms in manager.conf |
| 3:09AM |
0 |
Help! Directing Inbound calls to different extensions |
| 1:34AM |
2 |
Pickupexten not working |
| 1:01AM |
5 |
transfer calls via Manager Api |
| |
| Tuesday March 21 2006 |
| Time | Replies | Subject |
|---|
| 11:53PM |
1 |
Asterisk and gateway |
| 11:16PM |
3 |
PSTN to Asterisk VOIP in Manila |
| 10:05PM |
0 |
PRI not answering call after asterisk upgrade |
| 9:51PM |
5 |
Programming the Manager API |
| 9:44PM |
2 |
Multiple commands per priority |
| 9:35PM |
1 |
SIP video voicemail problem |
| 9:24PM |
2 |
TDM400 FXO module not answering or dialing out. |
| 9:02PM |
0 |
IAX2 issue with calls transfered between systems. |
| 8:32PM |
1 |
Cannot leave voicemail, Asterisk/Zaptel/libpi v1.0.9 |
| 5:01PM |
1 |
'Click to Dial' |
| 4:10PM |
0 |
Modem Bank |
| 3:57PM |
0 |
[OT] Cisco 7970 SCCP Image |
| 3:44PM |
4 |
ADT/Brinks alarm dialing through Asterisk |
| 3:35PM |
1 |
Polycom hand/head set echo and Zapata config |
| 3:17PM |
1 |
VoiceMailMain(@context) Problem with Option 5(Advanced) |
| 3:06PM |
0 |
Attachment Blocking Notification |
| 2:52PM |
2 |
VoiceMailMain(@context) Problem with Option 5 (Advanced) |
| 2:51PM |
12 |
Fw: anybody has SIP realtime working ? |
| 2:37PM |
1 |
Dimensions of TDM 24 card |
| 1:57PM |
6 |
FAX over PRI |
| 1:36PM |
2 |
callerid= in zapata.conf |
| 1:09PM |
0 |
Blind Transfer / Ring Groups Problem |
| 1:05PM |
2 |
Remote MWI over IAX2 |
| 12:53PM |
0 |
Sound dies |
| 11:31AM |
4 |
Realtime SIP Persistency |
| 11:24AM |
2 |
need to make my oh323 work with quintum no gatekeeper |
| 11:23AM |
0 |
Asterisk with TOPEX GSM Gateway |
| 11:19AM |
1 |
Problem with chan_iax.cimplimentationcausesbadaudio? |
| 11:07AM |
3 |
Realtime / SIP Peers etc |
| 11:00AM |
0 |
DTMF leak with IAXy & call waiting Bug? |
| 10:56AM |
1 |
ODBC and VoiceMail messages. |
| 10:48AM |
3 |
VoIP prepaid billing |
| 10:21AM |
0 |
SIP Realtime 1.2.5 and Username/auth name mismatch ? |
| 10:07AM |
5 |
Cisco POS 3-08-2 |
| 10:02AM |
3 |
WiFi phones and WDS (Wireless Distribution System) |
| 10:02AM |
6 |
Native MOH - Convert mp3 to ulaw |
| 9:44AM |
0 |
hfc-pci cards on ppc |
| 9:37AM |
7 |
Multiple processes |
| 9:19AM |
2 |
Problem with chan_iax.c implimentationcausesbadaudio? |
| 9:16AM |
4 |
Junghanns and Digium TDM400? |
| 8:55AM |
1 |
Problem with chan_iax.c implimentation causesbadaudio? |
| 8:51AM |
3 |
Zap<-->IAX codec? |
| 7:48AM |
1 |
Caller ID forwarding with Pickup() application? |
| 7:47AM |
1 |
Problem with chan_iax.c implimentation causesbad audio? |
| 7:46AM |
0 |
app_queue and ARA |
| 7:13AM |
2 |
Voice mail not working with Asteriks 1.2.5 |
| 7:09AM |
1 |
Web-ex type solution for use with asterisk |
| 2:53AM |
2 |
How to make groups of extensions ??? |
| 2:50AM |
2 |
How to make extension groups ??? |
| 2:00AM |
0 |
Queue and busy/congested ZAP channels |
| 1:34AM |
0 |
CDR problem with TAPI |
| |
| Monday March 20 2006 |
| Time | Replies | Subject |
|---|
| 4:47PM |
3 |
Problem with chan_iax.c implimentation causes bad audio? |
| 4:36PM |
0 |
problems with international dialing |
| 3:29PM |
1 |
Asterisk Disconnecting after 30sec when someone leaving VM |
| 2:28PM |
0 |
integration with Toshiba PBX system |
| 1:08PM |
0 |
Primary D-Channel on span 1 down |
| 12:57PM |
0 |
Experiences with ATA model Octtel SP200SO |
| 12:28PM |
2 |
Problem with intermittent one-way audio |
| 11:28AM |
0 |
sip show inuse not accurate |
| 11:18AM |
1 |
(no subject) |
| 11:07AM |
1 |
Is it possible to turn off password for transfers on FOP |
| 10:44AM |
1 |
Aterisk with Realtime |
| 10:04AM |
0 |
meetme recording very loud |
| 9:35AM |
0 |
assman, the ncurses asterisk manager interface |
| 9:21AM |
1 |
simple question on asterisk |
| 9:19AM |
2 |
pickup a call in queue |
| 9:14AM |
0 |
Prodding channel h323 |
| 8:52AM |
1 |
How often do YOU register? |
| 5:54AM |
1 |
answer delay |
| 5:24AM |
1 |
How to make caller groups ??? |
| 5:12AM |
0 |
How to setup Proxy info to * box , [* box behind a squid proxy and firewall ] |
| 4:53AM |
0 |
MixMonitor and transferred calls |
| 4:12AM |
0 |
Help: Using asterisk and mysql for a university project |
| 3:56AM |
4 |
simple perl-agi - where's the error? |
| 3:51AM |
2 |
ISDN Protocol Unknom Error with Junghanns OctoBRI |
| 3:14AM |
0 |
Problems loading res_odbc.so and cdr_odbc.so |
| 2:38AM |
3 |
Grabbing the billsec and duration after a hangup. |
| 2:32AM |
5 |
Numbered Voicemails even with delete option! |
| 1:32AM |
0 |
Using IAX |
| 1:16AM |
0 |
asterisk and DDI |
| 12:32AM |
1 |
计划生育的无耻宣传该结束了 |
| |
| Sunday March 19 2006 |
| Time | Replies | Subject |
|---|
| 11:41PM |
0 |
HFC USB (was MultiBRI in Australia - found one-maybe) |
| 9:42PM |
2 |
Local Channel |
| 8:30PM |
0 |
Xorcom TS-1 T1 installs? |
| 4:40PM |
1 |
HFC USB (was MultiBRI in Australia - found one - maybe) |
| 4:02PM |
0 |
Transfer to specific park number |
| 3:24PM |
2 |
Call Pickup Woes |
| 3:24PM |
0 |
ISDN NT Mode & CAPI |
| 3:04PM |
2 |
Grandstream unit HT-488 |
| 2:40PM |
0 |
Problem w/ Dial Command on Zap channel |
| 2:40PM |
1 |
trunking questions |
| 2:32PM |
0 |
Calls to SIP providers |
| 12:40PM |
2 |
Zaptel will not build |
| 11:39AM |
0 |
Voicemail Bug? |
| 10:57AM |
0 |
Bizzare DTMF on channel bank |
| 10:56AM |
3 |
Annoying Asterisk Realtime Limitation |
| 10:04AM |
0 |
Sending ANI to TDM40B FXS? |
| 7:41AM |
1 |
accessing speed dial database |
| 1:48AM |
0 |
PSTN lines permission settings to different extensions |
| 12:31AM |
7 |
An FXO version of IAXy? |
| 12:20AM |
3 |
g729 and latency measures |
| |
| Saturday March 18 2006 |
| Time | Replies | Subject |
|---|
| 8:32PM |
1 |
A general deployment question (OT) |
| 7:51PM |
1 |
Polycom IP600 - no ring? |
| 7:09PM |
0 |
Polycom IP600 dual ethernet port - bandwidth impact |
| 6:42PM |
1 |
GS BT102 dual ethernet port -bandwidth impact |
| 6:22PM |
0 |
Cisco 7960 dual ethernet port - bandwidth impact |
| 3:52PM |
0 |
Panasonic KX-TDA1000 with asterisk server |
| 12:21PM |
1 |
Realtime SIP users/peers - Screwed? |
| 12:16PM |
2 |
Jittery meetme conference using Linksys 942 phones |
| 11:54AM |
0 |
Realtime SIP users/peers |
| 11:09AM |
4 |
How to enable talking in chanspy while spying? |
| 9:42AM |
3 |
Sipura 3000 DMTF |
| 9:02AM |
0 |
I have my asterisk machine behind a Linux, Nat ... |
| 8:12AM |
1 |
List of transcoding combinations |
| 2:01AM |
0 |
Re: Server freeze with meetme and sip GSM users |
| 12:43AM |
0 |
I have my asterisk machine behind a Linux Nat ... |
| 12:03AM |
0 |
T38 Passthrough testing -- unknown media type error |
| |
| Friday March 17 2006 |
| Time | Replies | Subject |
|---|
| 10:57PM |
0 |
Question on compiling Zaptel |
| 10:54PM |
1 |
Re: DUNDi .... Halfway and CLUSTERING |
| 8:53PM |
0 |
Critical Problem with asterisk |
| 8:33PM |
11 |
Asterisk Users Mailing List Traffic |
| 6:11PM |
7 |
gsm picocells |
| 4:34PM |
3 |
SIP Realtime Users |
| 4:16PM |
0 |
New astGUIclient VICIDIAL Released: 1.1.10 |
| 3:21PM |
1 |
One-Way SIP Audio with SVN Codebase (CANCEL) |
| 2:43PM |
0 |
OT: Good Vendor? |
| 2:29PM |
0 |
FreePBX 2.0.1 released! |
| 2:23PM |
3 |
Exchange 12 Unified Messaging |
| 2:12PM |
1 |
Re: DUNDi .... Halfway and CLUSTERING |
| 1:34PM |
1 |
More Voicemail prompts |
| 1:15PM |
1 |
Re: DUNDi .... Halfway and CLUSTERING |
| 12:45PM |
1 |
how to get separate CDR for inbound and outbound legs of a call |
| 12:40PM |
0 |
Call transfer problems, SOLVED |
| 12:39PM |
1 |
Extra Debugging without console |
| 12:37PM |
2 |
IAX Phone? |
| 12:13PM |
0 |
How to set priority for SJPhone. |
| 10:33AM |
1 |
problem with tdm22b |
| 10:32AM |
0 |
Transfer problems revisited |
| 10:11AM |
4 |
Aastra Questions |
| 9:47AM |
6 |
Disappearing voicemail |
| 9:38AM |
0 |
caller unable to transfer |
| 9:08AM |
1 |
Sticky Problem SER/Asterisk |
| 9:00AM |
0 |
CRM + Phones |
| 8:55AM |
4 |
Countries supporting SMS on PSTN (ISDN) |
| 8:16AM |
0 |
Asterisk and PacketCable |
| 8:05AM |
7 |
problems with emailing voicemail |
| 7:55AM |
2 |
DISA alternative |
| 7:36AM |
1 |
RE: DUNDi .... Halfway and CLUSTERING |
| 7:13AM |
2 |
choppy recorded sounds in asterisk |
| 7:10AM |
1 |
Asterilink?!?! |
| 6:59AM |
0 |
(no subject) |
| 6:47AM |
4 |
TDM 2400 With 24 FXO |
| 6:46AM |
4 |
D4 AMI - No Caller ID |
| 6:42AM |
2 |
Analog POTS line -> Rhino FXO Channel Bank -> No Hangup |
| 6:31AM |
0 |
Set CallerID to a specific Queue Member |
| 6:27AM |
0 |
[FOLLOWUP]: Calls not tearing down properly |
| 6:19AM |
0 |
Keeping the user name in sip INVITE with fixed IP host routing. |
| 6:07AM |
1 |
Asterisk on hosted server |
| 6:03AM |
0 |
How to install the cdr_odbc module. |
| 5:11AM |
3 |
Best budget IP phone at the moment? |
| 4:55AM |
0 |
Call pickup between different protocols |
| 4:15AM |
0 |
One-Way SIP Audio with SVN Codebase |
| 4:12AM |
2 |
asterisk and skype - asterisk newbie |
| 3:03AM |
3 |
TFTP problems on FC4 |
| 2:45AM |
2 |
embedded hardware for Asterisk? |
| 2:13AM |
1 |
automatic fax detection in asteriskathome |
| 1:49AM |
0 |
OT: reset LinkSys 941 to factory defaults & howto config' via TFTP |
| 1:48AM |
3 |
Numbered Voicemails when you still delete them. |
| 1:43AM |
0 |
asterisk configurations |
| 1:34AM |
0 |
Echo/Milliwatt Test Numbers in Oz ? |
| 1:29AM |
0 |
OT: any one in Stockholm for quick piece of advice |
| 1:16AM |
1 |
french sounds in asterisk |
| |
| Thursday March 16 2006 |
| Time | Replies | Subject |
|---|
| 10:30PM |
1 |
Feedback from VON expo!Infoon*HAandPolycomphone!! |
| 10:27PM |
0 |
RE: DUNDi .... Halfway and CLUSTERING |
| 10:25PM |
3 |
Question about advanced IVR |
| 10:23PM |
0 |
qozap drops -- possible to bridge BRIstuff ISDN to analog zaptel phone? |
| 8:50PM |
0 |
Feedback from VON expo! Infoon*HAandPolycomphone!! |
| 7:38PM |
7 |
OT: Unblocking bloced CID |
| 5:25PM |
0 |
Creating Asterisk Bounties |
| 5:24PM |
1 |
DUNDi .... Halfway |
| 5:17PM |
0 |
Small noise every 3 seconds |
| 4:36PM |
2 |
Queues Not Reporting Estimated Hold Time |
| 4:28PM |
1 |
UK Caller ID - Asterisk 1.2.5 - TDM4 Card |
| 4:28PM |
4 |
New one on me: How to UN-transfer |
| 4:19PM |
0 |
redirect output |
| 3:58PM |
0 |
(no subject) |
| 3:48PM |
1 |
Newbie needs audio help |
| 3:37PM |
0 |
Implementing VoIP for first time with Packetcable |
| 3:30PM |
3 |
voip-info.... again |
| 3:24PM |
0 |
SCCP problem with ATA188, Asterisk@home and chan_sccp |
| 3:14PM |
0 |
Two dissimilar data only T1 lines possible in Asterisk? |
| 3:10PM |
0 |
Regcontext, only 1 context available? |
| 2:46PM |
1 |
Authenticate CDR Logging |
| 2:38PM |
1 |
G.729 codec licencing |
| 2:35PM |
0 |
3 way calls & transfers |
| 1:42PM |
0 |
Asterisk select outbound trunk based on minutes used per month?? |
| 1:36PM |
1 |
RFC 2833 and SIP? DTMF? What am I not getting? |
| 1:34PM |
1 |
MFCR2 |
| 1:26PM |
1 |
Re: transfers/parked calls + polycom 501 |
| 12:03PM |
3 |
Feedback from VON expo! Info on * HA and Polycomphone!! |
| 11:54AM |
2 |
Can anybody get me setup with a hosted asterisk@home box or virtual server in the next 24 hours? |
| 11:40AM |
1 |
Re: transfers/parked calls + polycom 501 |
| 11:36AM |
0 |
Budgetone strange problem - have to press hold on and off to connect call. |
| 10:45AM |
1 |
capiHOLD missing in BRIstuff 0.3.0 |
| 10:31AM |
0 |
Dialplan : Forwarding call to voicemail after onering iif extension is busy |
| 9:45AM |
0 |
Dialplan : Forwarding call to voicemail after one ring iif extension is busy |
| 9:35AM |
1 |
MeetMe - Causes * to crash :/ |
| 9:35AM |
1 |
Creating a voip network... use asterisk? |
| 9:30AM |
0 |
Zap channel not hanging up |
| 9:29AM |
0 |
Asterisk Users Group Tonight, Irvine, Ca |
| 9:24AM |
1 |
setting callerid not working if no callerid on incoming number |
| 9:09AM |
0 |
Testing IAX links |
| 8:58AM |
0 |
Fw: help required configuring card |
| 8:55AM |
1 |
Feedback from VON expo! Info on *HAandPolycomphone!! |
| 8:45AM |
0 |
Feedback from VON expo! Info on * HAandPolycomphone!! |
| 8:30AM |
1 |
Attended call transfer with GXP-2000 |
| 8:20AM |
2 |
Feedback from VON expo! Info on * HA andPolycomphone!! |
| 8:16AM |
0 |
ODBC voicemail storage |
| 8:09AM |
1 |
module load order for Junghanns qozap and TDM card |
| 7:52AM |
1 |
ISDN BRI and UK Premium Rate Numbers |
| 7:29AM |
1 |
Queues - calls going to agents lised as "In use" |
| 7:05AM |
0 |
can't get TDM400P to answer |
| 6:06AM |
0 |
How to configure PSTN lines permissions todifferent extensions ??? |
| 5:11AM |
1 |
open source queue analyzer |
| 4:34AM |
0 |
[Fwd: Re: Sync Source: Internally clocked] |
| 4:24AM |
1 |
Codecs? - Asked to transmit frame type 256, while native formats is 8 (read/write = 8/8) |
| 4:04AM |
4 |
asterisk@home V's Asterisk |
| 3:02AM |
0 |
Server freeze with meetme and sip GSM users and ztdummy |
| 2:09AM |
2 |
SER & Asterisk with DID incoming and out going |
| 1:50AM |
1 |
asteriskathome maximun channels per trunk |
| 1:48AM |
0 |
carry forward uniqueid |
| 1:40AM |
2 |
SIP routing over IAX2 |
| 1:05AM |
4 |
How to transmit Video |
| 1:04AM |
0 |
Wanted: IAX ATA w/ FXO |
| 12:55AM |
1 |
send text to a device |
| 12:07AM |
2 |
Problem with System() command. |
| |
| Wednesday March 15 2006 |
| Time | Replies | Subject |
|---|
| 10:11PM |
2 |
How to configure PSTN lines permissions to different extensions ??? |
| 8:55PM |
0 |
FXS Caller ID? |
| 8:07PM |
2 |
GUI Web interface |
| 6:04PM |
1 |
dropping voice frame ulaw - slin? |
| 5:42PM |
2 |
Do Not Disturb? |
| 3:35PM |
3 |
Failed to read gains: Invalid argument |
| 3:07PM |
1 |
Unable to forward frame |
| 3:01PM |
0 |
sporadic voicemail delete problems |
| 2:46PM |
3 |
Echo canceller data-points |
| 2:26PM |
2 |
Speeding up the dial of DTMF's in SIP channel |
| 2:10PM |
0 |
T.38 Passthrough testing -- IAX problem |
| 2:04PM |
0 |
RE: Asterisk-Users Digest, Vol 18, Issue 147 |
| 1:45PM |
3 |
Sync Source: Internally clocked |
| 1:45PM |
0 |
(no subject) |
| 1:12PM |
0 |
(no subject) |
| 1:06PM |
2 |
Help with Gizmo from outside firewall |
| 12:48PM |
0 |
definity prologix |
| 11:58AM |
0 |
OT: Using Sipsak to reboot a Snom phone < -a nswered my own question |
| 11:48AM |
0 |
Call go on hold for no reason |
| 11:46AM |
0 |
Idiot's guide to Q.932? |
| 11:40AM |
0 |
OT: Using Sipsak to reboot a Snom phone |
| 10:33AM |
5 |
how to show called name on calling polycomdisplay |
| 10:33AM |
2 |
Script to Restart Zaptel |
| 10:15AM |
0 |
openSUSE 10.0 and zaptel init script |
| 10:05AM |
3 |
Double-ring tone |
| 9:44AM |
1 |
ooh323 Gatekeeper Bug |
| 9:25AM |
2 |
Fake Ring Tone/Compile Addon |
| 9:21AM |
0 |
res_config_mysql.so not found |
| 8:50AM |
2 |
Asterisk integration with office PBX |
| 8:42AM |
1 |
cisco 7912 not taking config |
| 8:39AM |
1 |
cards |
| 8:37AM |
9 |
OSHA requirement to "reach a live human" ?? |
| 8:17AM |
0 |
Aastra 480i CT - multiple lines? |
| 8:04AM |
0 |
Re: Stuck. Extenions.conf? Realtime? MySQL? |
| 8:01AM |
1 |
Toshiba Strata DK-280 support? |
| 7:58AM |
1 |
Development news :: T38 passthrough |
| 7:44AM |
0 |
Attended transfers timing out after 3 rings |
| 7:41AM |
3 |
problem configuring a digium quad E1 card |
| 7:39AM |
0 |
Zaptel compile errors on x86_64 - DEFINE_SPINLOCK??? |
| 7:24AM |
0 |
Meetme monitoring only bug |
| 7:15AM |
1 |
external modem |
| 6:37AM |
4 |
misdn problem |
| 6:33AM |
1 |
AVM C2 chan_capi-cm-0.6.3 Error on Dial |
| 5:41AM |
0 |
IMACS800 |
| 5:20AM |
1 |
asterisk crash too much? |
| 4:13AM |
0 |
[SPAM] [asterisk-dev] CALL FOR COMMENTS - Dialplan |
| 3:44AM |
2 |
(unexplicable) peaks of machine load |
| 3:23AM |
0 |
spa 3000/2100 noise |
| 3:23AM |
0 |
there is lack behind in recoded calls via sox |
| 3:19AM |
0 |
There is lacking behind in recorded calls via sox |
| 3:14AM |
1 |
asterisk perl commands |
| 2:52AM |
6 |
Cisco phones and Linksys SRW224P |
| 2:49AM |
3 |
Zaptel compile errors on x86_64 |
| 2:29AM |
0 |
MCC v.1.3 Released |
| 2:08AM |
1 |
How to assign a specific PSTN line to a specific extension ??? |
| 1:52AM |
3 |
how to show called name on calling polycom display |
| 1:37AM |
1 |
IVR weirdness |
| 12:32AM |
3 |
Asterisk to receive fax |
| |
| Tuesday March 14 2006 |
| Time | Replies | Subject |
|---|
| 11:16PM |
1 |
invalid wav gsm frame size: 1 bytes ?? |
| 10:57PM |
2 |
asterisk and iptables |
| 10:53PM |
1 |
RE: Problems with installing a TE110P on a Dell Poweredge 850 |
| 9:51PM |
5 |
Asterisk Native Sounds - in case you missed it... |
| 9:30PM |
1 |
Bug Help or Suggestion - Grandstream GXP2000 (firmware 1.0.2.8) - BLF, Hints, call-limit |
| 9:23PM |
1 |
Adding entries on company directory |
| 9:02PM |
4 |
Stuck. Extenions.conf? Realtime? MySQL? Grrrrr! |
| 9:01PM |
1 |
asterisk hang |
| 8:01PM |
2 |
isdn out of band signalling |
| 6:41PM |
1 |
Directory doesn't work well Asterisk@home2.7- try from PSTN with Digital recepcionist- Directory based on Last name |
| 6:13PM |
2 |
Max retries exceeded to host... |
| 5:32PM |
1 |
Problems with installing a TE110P on a Dell Poweredge 850 running Fedora Core 4 |
| 5:26PM |
0 |
help required configuring card |
| 4:16PM |
5 |
New ncurses Asterisk Manager Interface |
| 4:13PM |
0 |
External transfer |
| 3:24PM |
0 |
ANNOUNCEMENT : A2Billing (Asterisk2Billing) - release v1.1 |
| 2:26PM |
0 |
invoking a macro doesn't work |
| 2:15PM |
0 |
LNP / DID Service - Louisianna / Virginia |
| 1:46PM |
1 |
E911 from Remote Office via PRI |
| 1:41PM |
0 |
LCDPROC cient for Asterisk |
| 1:34PM |
0 |
ip telephony project |
| 1:34PM |
0 |
Flash on Unicall Channel |
| 12:34PM |
7 |
Realtime Extensions |
| 11:50AM |
3 |
Outbound paging dialplan example? |
| 11:47AM |
0 |
MWI & Asterisk Realtime Architecture |
| 11:34AM |
3 |
EICON Diva 4BRI |
| 11:22AM |
3 |
Voice volume using Monitor application |
| 11:14AM |
1 |
channel bridging |
| 11:08AM |
0 |
List Rules |
| 10:52AM |
2 |
OT - force Cisco phones to reboot |
| 10:38AM |
3 |
Attended Transfer - transfer timeout, how to change? |
| 10:18AM |
1 |
Codec Issue |
| 10:10AM |
0 |
Asterisk Users Group Meeting March 16, Irvine, Ca |
| 9:44AM |
6 |
IAX choppy sound |
| 9:27AM |
1 |
Bad FXS Module? |
| 9:21AM |
0 |
Asterish Guru needed in Phoenix ASAP |
| 9:01AM |
1 |
Re: Asterisk-Users Digest, Vol 20, Issue 91 |
| 8:48AM |
2 |
digium.com redesign |
| 8:31AM |
0 |
Problem with uac_replace and corrupted From |
| 8:12AM |
2 |
Realtime SIP |
| 7:57AM |
0 |
Sample SER + Asterisk conf? |
| 7:43AM |
0 |
[OT?]SCCP image for cisco 7905g |
| 7:08AM |
0 |
Problem with poud key (#) |
| 6:59AM |
0 |
Line connections |
| 6:34AM |
1 |
10minutes to restart Asterisk@home 2.7 |
| 5:59AM |
2 |
Latest Dell SC430 Compatibility With Wildcard |
| 2:31AM |
0 |
DATA CALLS annoying my system |
| 2:19AM |
0 |
Inbound sipgate number forwarding to differnet users |
| 1:01AM |
0 |
I can't resume a call on hold from zap device |
| |
| Monday March 13 2006 |
| Time | Replies | Subject |
|---|
| 7:27PM |
0 |
Call Parking Grandstream |
| 7:13PM |
0 |
Spam? Re: Unknown signalling method 'pri_cpe' |
| 7:11PM |
7 |
Clustering "NEW THREAD", Almost Working |
| 6:47PM |
2 |
Unknown signalling method 'pri_cpe' |
| 6:32PM |
2 |
CDR Bug? |
| 5:32PM |
2 |
Can One FXO Support Multiple Phone Lines? |
| 5:14PM |
2 |
Simple php script to monitor asterisk calls |
| 5:13PM |
2 |
DISA & SPA3000 issues |
| 5:10PM |
1 |
incoming limit, call_limit, or call-limit? |
| 4:12PM |
1 |
MWI to 7960's sometimes delayed or lost. Please advise. |
| 3:32PM |
0 |
(no subject) |
| 3:27PM |
4 |
slinear bandwidth |
| 2:52PM |
1 |
Seperate music on hold for SIP extensions |
| 2:36PM |
1 |
cisco 7912 ringlist.dat file format |
| 2:31PM |
0 |
Phase locked mode |
| 2:20PM |
0 |
Calls not tearing down properly |
| 1:59PM |
1 |
SIP Jitter Buffer for 1.2.5 |
| 1:42PM |
1 |
All calls in queue go to agent that is down?? |
| 1:39PM |
2 |
Dumb question (hang up detection/Zapata.conf) |
| 1:14PM |
5 |
Cisco 7960 8.2 callerID lists proxy? |
| 12:47PM |
1 |
Outgoing calls via Sipgate |
| 12:22PM |
0 |
Re: Regexten & Regcontext, working now |
| 12:19PM |
4 |
priorityjumping=no |
| 12:18PM |
0 |
chan_zap ast_pickup_call issue redux |
| 11:28AM |
1 |
Spam? Re: Failed installing zaptel |
| 11:15AM |
0 |
Asterisk end Festival |
| 11:01AM |
0 |
How to forward inbound sipgate calls to different users in my entreprise, ( |
| 10:33AM |
1 |
Zaptel not compiling on lastest Centos 4.2 kernel. |
| 10:31AM |
0 |
Re: transfers/parked calls + polycom 501 |
| 10:27AM |
0 |
Regexten & Regcontext |
| 10:25AM |
1 |
Considering Asterisk |
| 10:22AM |
1 |
Failed installing zaptel |
| 10:20AM |
1 |
Hardware timing source for MeetMe |
| 9:58AM |
0 |
channel manipulation |
| 9:40AM |
1 |
music on hold without mpg123 |
| 9:38AM |
2 |
Diff between X100M and X100P? |
| 9:37AM |
1 |
Avaya IP Office 412 |
| 8:59AM |
1 |
Cannot load wcfxo -- Please help! |
| 8:47AM |
0 |
Incoming Call keeps ringing when the second call arrives |
| 8:35AM |
1 |
misdn |
| 8:20AM |
2 |
Re: transfers/parked calls + polycom 501 |
| 8:06AM |
2 |
Asterisk large scale, help needed |
| 7:55AM |
5 |
echo problem + choppy sound |
| 7:55AM |
1 |
Need help implementing call center featuresofAsterisk |
| 6:49AM |
1 |
Scrolling messages |
| 5:48AM |
1 |
Need help implementing call center features ofAsterisk |
| 5:25AM |
2 |
Need help implementing call center features of Asterisk |
| 3:51AM |
1 |
G729A |
| 2:04AM |
3 |
Callerid on transfer |
| 12:52AM |
1 |
Asterisk RealTime Question, Please help |
| 12:19AM |
1 |
Australian approved 4BRI PCI adapter preliminarytesting results |
| |
| Sunday March 12 2006 |
| Time | Replies | Subject |
|---|
| 11:13PM |
1 |
Action on phone pickup |
| 10:54PM |
3 |
Multiple IAX clients behind a firewall |
| 10:21PM |
7 |
stop monitor on transfer |
| 10:05PM |
0 |
Invitation |
| 7:00PM |
1 |
Australian approved 4BRI PCI adapter preliminary testing results |
| 5:00PM |
1 |
Flash zap trunk from Softphone or IP Handsets... |
| 3:43PM |
1 |
Hung IAX Channels |
| 1:32PM |
0 |
DTMF digits lost on TDM400 card |
| 12:08PM |
1 |
Looking for docs on adjusting txgain/rxgain |
| 12:08PM |
1 |
Understanding queue timeouts + possible bug found |
| 11:27AM |
1 |
Building a small Office EPABX with VoIP GW with Asterisk |
| 8:17AM |
1 |
interop problem: "Missing handling for mandatory IE 24 (cs0, Channel Identification)" |
| 6:33AM |
1 |
Calls from PSTN , answering, When transfered get Hungup 'Zap/1-1' |
| 5:19AM |
1 |
Call and then play IVR |
| 4:58AM |
4 |
Voice problem |
| 2:31AM |
1 |
Speakeasy VOIP + Asterisk? |
| |
| Saturday March 11 2006 |
| Time | Replies | Subject |
|---|
| 8:57PM |
0 |
how to check if ztdummy is working properly? |
| 4:46PM |
4 |
Polycom - directory dial |
| 4:21PM |
1 |
Autodial |
| 2:13PM |
0 |
Unicall and Fax detection |
| 2:11PM |
1 |
Limiting the number of concurrent calls for a group of SIP devices |
| 12:53PM |
1 |
OT: Flash/web site developer in Boca Raton FL required |
| 12:37PM |
0 |
I don't listen first seconds of audio from call - Asterisk integration with old PBX |
| 9:23AM |
0 |
Incompatible switchtypes |
| 9:04AM |
1 |
how to connect 3 or more servers via IAX ? |
| 8:53AM |
0 |
Clustering / Dundi |
| 5:16AM |
1 |
FW: I need to set NO CRC4 on zaptel.conf? |
| 4:24AM |
1 |
HITBSecConf2006 - Malaysia: Call for Papers |
| 4:09AM |
2 |
IVR dial by extension option.. |
| 1:26AM |
1 |
hotel vmail and iax trouble |
| 1:13AM |
0 |
asterisk having problem in playing sounds |
| 1:01AM |
0 |
Odd CID issue calling SIP to SIP DID - anyone have this or can explain it? |
| |
| Friday March 10 2006 |
| Time | Replies | Subject |
|---|
| 9:29PM |
0 |
change voicemail folders |
| 8:56PM |
1 |
(no subject) |
| 3:35PM |
1 |
voicetronix and asterisk@home |
| 3:28PM |
0 |
FW: T1 card Ali |
| 2:54PM |
3 |
Development news :: T38 passthrough support |
| 2:34PM |
4 |
Analog Desktop Phone |
| 2:25PM |
1 |
IAX / Firefly handshake problem |
| 2:08PM |
3 |
RFC Follow Me Find Me script |
| 2:07PM |
1 |
Junghanns, Germany ISDN settings |
| 2:01PM |
2 |
Problem compiling zaptel on latest RHEL kernel (2.6.9-34.EL) |
| 1:57PM |
4 |
dipura 2002 auto dial or intercom |
| 1:41PM |
1 |
HOWTO initialize new kernel & kernel source without reboot |
| 1:19PM |
8 |
Asterisk programmer needed |
| 1:14PM |
0 |
Voice Mail woe |
| 12:57PM |
3 |
Menu in queue |
| 12:22PM |
27 |
Clustering |
| 12:20PM |
1 |
cidname via IAX2? |
| 12:20PM |
0 |
Voicetronix OpenSwitch / Sangoma Analog Card |
| 11:04AM |
1 |
Yet again: chan_zap.c: Unable to specify channel 4: No such device |
| 10:52AM |
2 |
7970 Configs |
| 10:19AM |
2 |
Disable flash transfers? |
| 10:06AM |
0 |
queue and service period |
| 9:55AM |
0 |
Background timeout and Read questions |
| 9:45AM |
1 |
ADPCM - vs - G.726 |
| 9:39AM |
0 |
TDM400 DTMF Caller ID |
| 9:29AM |
2 |
Action after _caller_ has hungup(cmd Dial 'g'-option) |
| 9:04AM |
0 |
RE: Stable Hardware Combination Experiences |
| 8:47AM |
0 |
DUNDi Public and Private Key Question |
| 8:32AM |
1 |
ring (hunt?) group |
| 7:57AM |
2 |
Dial plans and forwarded phones |
| 7:51AM |
0 |
Forward from SER to asterisk can't hang up |
| 7:46AM |
0 |
pstn to asterisk, DVG-3004S, MP104? |
| 7:29AM |
0 |
Operator consoles for large systems |
| 7:09AM |
0 |
Sangoma A101 T1/E1 (PRI) voip card available for testing |
| 7:04AM |
4 |
difference between records in CDR and real duration of call |
| 6:15AM |
2 |
IAX2 + Sonicwall |
| 5:48AM |
2 |
ZT_CHANCONFIG failed on channel 2: , Guidance requested |
| 5:25AM |
0 |
Flash call transfer problem |
| 4:22AM |
3 |
Dial Out IVR |
| 3:20AM |
0 |
ALSA channel (console/dsp) problem |
| 2:59AM |
1 |
monitor/statistic web interface for cdr |
| 2:00AM |
1 |
Can I avoid configuring FXS part in zaptel.conf and zapata.conf |
| 1:50AM |
0 |
mysql asterik |
| 1:49AM |
1 |
7960 Cisco SIP Phone TFTP Files |
| 1:39AM |
0 |
mediatrix 1102 |
| 1:05AM |
1 |
Configs for Gradwell and inWeb |
| 12:44AM |
1 |
Extensions base policy |
| |
| Thursday March 9 2006 |
| Time | Replies | Subject |
|---|
| 11:08PM |
0 |
PRI/T-1 |
| 10:30PM |
1 |
Asterisk Re-invites - how to tell ? |
| 9:50PM |
2 |
How to assign channels for asterisk |
| 8:32PM |
3 |
OT: Snom 320, displaying text on the scree n from * |
| 8:22PM |
0 |
Not getting mails from Mar 2 |
| 8:04PM |
2 |
Sangoma A200 error |
| 7:21PM |
0 |
Mitel SX-2000 <--> TE210P Red Alarm |
| 4:37PM |
2 |
OT: Snom 320, displaying text on the screen from * |
| 3:15PM |
2 |
Extracting info from the $EXTEN variable |
| 3:05PM |
1 |
Polycom 4000 results? |
| 2:43PM |
0 |
Nortel BCM and Asterisk as SIP Extension |
| 2:30PM |
1 |
SIP/Video client for PocketPC that works with Asterisk? |
| 2:00PM |
4 |
IVR woes |
| 1:50PM |
1 |
Single E1 with HW Echo Can? |
| 1:40PM |
0 |
best pre paid for astreisk? |
| 12:37PM |
1 |
G729, G729 annex A or G729 annex B? |
| 12:19PM |
1 |
news-reading question |
| 11:55AM |
1 |
Chinaroby VOIP phones? SECOND TIME! |
| 11:20AM |
0 |
T1 card Ali |
| 10:46AM |
0 |
AMD64 x2 and asterisk 1.2.4 not hearing demo-congrats |
| 10:27AM |
2 |
RES: DTFM or FSK |
| 10:16AM |
1 |
Oneway voice |
| 10:01AM |
0 |
res_musiconhold.c: Only wrote -1 of 640 bytes to pipe // no queue music |
| 9:16AM |
1 |
SPA3000 and callerID |
| 9:06AM |
0 |
broken pipe, restart asterisk |
| 8:59AM |
2 |
TDM11B Hang up detection not working in France ? |
| 8:46AM |
2 |
Merlin Magix Integration |
| 8:20AM |
1 |
Getting to the last "old" voicemail message |
| 8:09AM |
3 |
DTFM or FSK |
| 8:04AM |
3 |
cdr data |
| 7:44AM |
0 |
Stress Tests from AsteriskGur with Asterisk@Home |
| 7:39AM |
0 |
paly sound when we Start and stop recording |
| 7:26AM |
1 |
Jitter buffer for SIP channels (OT?) |
| 7:05AM |
2 |
7940/60 SIP 8.2 |
| 7:00AM |
1 |
Is extension.conf documentation wrong? |
| 6:49AM |
1 |
digium certification for Europe |
| 6:28AM |
0 |
ruby-agi-1.1.2 released |
| 6:17AM |
0 |
Re: [asterisk-biz] Professional Recordings |
| 6:00AM |
0 |
Music On Hold playback |
| 5:23AM |
0 |
Fax behind ATA |
| 4:43AM |
5 |
Festival tts |
| 3:17AM |
1 |
Asterisk code help |
| 2:00AM |
0 |
Attended transfer returns invalid extension |
| 1:46AM |
0 |
Can't hear busy tone |
| 12:57AM |
0 |
Does Atcom AU-200 work with XLite? |
| |
| Wednesday March 8 2006 |
| Time | Replies | Subject |
|---|
| 10:53PM |
0 |
re: Billing Package for Asterisk |
| 8:00PM |
0 |
Openline4 and asterisk@home |
| 5:36PM |
0 |
MINNESOTA: TwinCities Asterisk Users Group - Saturday 03/11/2006 |
| 4:45PM |
0 |
2-Asterisk@Home Servers Connecting Portugal to Brazil (offices) |
| 4:22PM |
0 |
overlap dialing with polycom? |
| 4:19PM |
6 |
Professional Recordings |
| 4:17PM |
0 |
Unicall, Fax and Echo cancellation |
| 4:05PM |
1 |
Re: PLEASE respond: how to get Asterisk to change coders on RTP handoff?? HELLO??? |
| 3:42PM |
0 |
Random Zap port going crazy When channel released after a flash. |
| 2:30PM |
1 |
Any way to change dns timeout value? Asterisk hangs if internet unreachable |
| 2:14PM |
0 |
List Problems |
| 1:49PM |
0 |
More 7940 Questions |
| 1:43PM |
5 |
Cisco 7960 SIP - Displaying Time |
| 12:32PM |
1 |
Zap not installing |
| 12:17PM |
0 |
Faxing with MFC/r2 |
| 12:16PM |
0 |
Softphone for Windows CE 3.0 |
| 12:07PM |
3 |
No DTMF |
| 11:48AM |
1 |
Memory Problems |
| 11:12AM |
1 |
impact of qualify=yes |
| 10:53AM |
2 |
Putting caller in queue and dialing an extension simultaneously |
| 10:51AM |
0 |
RES: Inserting access codes as prefixes to CID |
| 10:51AM |
1 |
What port mpg123 uses for MoH? |
| 10:47AM |
1 |
Upgrading Asterisk witk G729 license installed |
| 10:43AM |
3 |
RES: pap2 Dial plan |
| 10:43AM |
4 |
PAP2 won't make two g729 calls at the same time |
| 10:23AM |
1 |
Location of MeetMe Recordings |
| 9:54AM |
0 |
pickup last ringing phone |
| 9:26AM |
0 |
Cisco Call Manager SIP trunk + Asterisk |
| 9:14AM |
4 |
Is everyone getting mails except me? |
| 9:09AM |
0 |
Chinaroby VOIP phones? |
| 8:20AM |
0 |
status on jitter buffer for SIP/RTP? (OT?) |
| 8:19AM |
1 |
Asterisk @ Home 2.6 Call hangs up |
| 7:38AM |
1 |
Asterisk sip and radius authentication |
| 7:19AM |
2 |
parking slot lights - testers wanted |
| 6:26AM |
1 |
[Slightly OT] Does TE110P (a 32-bit PCI) fit into PCIe x8 slot? |
| 6:19AM |
0 |
Mitel SX-2000 and Asterisk integration |
| 5:11AM |
1 |
Calls forwarding to numbers only in user's context |
| 5:00AM |
0 |
Clock is runing too fast, Asterisk@home2.5 Ztdummy and VMware workstation |
| 4:22AM |
0 |
Conference room owner Changing his room password? Ast@Home |
| 4:15AM |
0 |
Size'ing/performance |
| 3:47AM |
0 |
can't call some numbers/providers , ani code missing in sip header (found the problem, not the solution) |
| 3:17AM |
2 |
REGISTER headers changed |
| 1:15AM |
4 |
sending text to display of sip phones |
| 12:49AM |
0 |
Hangup with error |
| |
| Tuesday March 7 2006 |
| Time | Replies | Subject |
|---|
| 11:52PM |
0 |
Called number not recognised |
| 10:46PM |
0 |
icmp 36: 192.168.30.32 udp port 5004 unreachable |
| 10:16PM |
1 |
MeetMe 'i' option not working correctly? |
| 8:02PM |
1 |
OT: Polycom Registration Weirdness |
| 7:42PM |
0 |
Mitel SIP firmware |
| 7:03PM |
3 |
Reverse group in zapata.conf |
| 5:54PM |
0 |
HOWTO volume per (7960) phone |
| 5:34PM |
0 |
IAXy (S101) echo? |
| 5:13PM |
0 |
Agents and agent counts |
| 4:26PM |
6 |
System Design |
| 4:20PM |
2 |
OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3 |
| 3:58PM |
3 |
Re: [asterisk-dev] Is there a way to define an outbound proxy in sip.conf ? |
| 2:37PM |
0 |
Asterisk Compatible Server Architecture |
| 2:36PM |
5 |
MWI, SER and asterisk |
| 1:08PM |
5 |
can i get the script |
| 12:53PM |
1 |
Question from a newbie on finding digium hosts |
| 11:34AM |
1 |
Changing REINVITE status of the channel dynamically |
| 11:33AM |
5 |
Receiving Multiple calls on asterisk at home |
| 11:22AM |
1 |
Call Path Optimization? |
| 11:21AM |
1 |
Setting Vaaibles |
| 10:56AM |
2 |
Using softphone from a remote location to get into * |
| 10:53AM |
1 |
OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3 WARNING!!! |
| 10:46AM |
2 |
pap2 Dial plan |
| 10:39AM |
0 |
anonymous caller id causes crash |
| 9:53AM |
7 |
res_mysql.conf & DNS SRV lookup |
| 9:31AM |
1 |
I can't receive multiple pages with spandsp |
| 9:28AM |
1 |
PLEASE HELP ,a2billing problem with call duration |
| 9:12AM |
9 |
Oh this is bad.... bindaddr and rtp traffic |
| 9:00AM |
3 |
indications & SIP |
| 8:58AM |
0 |
Toll free nos |
| 8:49AM |
0 |
Destroying a SIP extension doesn't destroyvoicemail box?is this a bug? |
| 8:15AM |
1 |
Asterisk + SE Linux |
| 8:02AM |
1 |
PBX-VPN-SIP-Asterisk trouble |
| 8:02AM |
0 |
Destroying a SIP extension doesn't destroy voicemail box?is this a bug? |
| 7:37AM |
2 |
Send One Touch Record to mail |
| 7:32AM |
0 |
a2billing problem with call duration |
| 6:57AM |
3 |
Problem ChanSpy |
| 6:40AM |
3 |
How to change Budgetone dialtone? |
| 6:16AM |
0 |
webvmail problems |
| 4:41AM |
0 |
Two Asterisk server |
| 4:08AM |
1 |
Help! Connecting two Astersik via SIP channels |
| 3:01AM |
2 |
Asterisk Prepaid Card |
| 3:01AM |
6 |
ON DEMAND call Recording |
| 2:20AM |
2 |
Periodic-announce in queues |
| 1:40AM |
0 |
Gmane - Asterisk Users Mailing List |
| 1:25AM |
0 |
Asterisk add-ons - H323 |
| 1:11AM |
0 |
R: Capturing DTMF during a call |
| |
| Monday March 6 2006 |
| Time | Replies | Subject |
|---|
| 11:32PM |
3 |
What is asterisk |
| 10:29PM |
1 |
Hangup issues |
| 8:20PM |
4 |
Asterisk download file locations |
| 7:15PM |
2 |
Confusion about construction of RURIs from contact headers for BYEs generated by * |
| 6:45PM |
1 |
ENUM lookup issues with e164.org |
| 6:18PM |
0 |
streaming recordings |
| 5:19PM |
1 |
PRI CID signalling not working? |
| 4:04PM |
2 |
Comedian Mail Add-ons? |
| 3:57PM |
1 |
most common VOIP echo simulaton for research purposes ? |
| 3:39PM |
1 |
IPv6 |
| 3:32PM |
0 |
Asterisk-addons 1.2.2 released |
| 2:25PM |
2 |
Problem getting two x200p cards working on 1.2.4 |
| 2:00PM |
3 |
call manager integration |
| 1:43PM |
0 |
Music on hold volume too high - using built in music on hold. |
| 1:00PM |
1 |
Asterisk and CISCO 7970 color |
| 12:16PM |
0 |
No ring when doing blind transfer. |
| 11:34AM |
1 |
Upgrading AAH |
| 11:24AM |
0 |
Initiate and monitor multiple calls? |
| 11:11AM |
1 |
PLEASE respond: how to get Asterisk to change coders on RTP handoff?? |
| 10:55AM |
0 |
call files and cdr I need src different from CallerID(number) |
| 10:53AM |
1 |
cdr records on transfer |
| 10:53AM |
2 |
Polycom voice.gain.tx.analog.handset and asterisk echo |
| 10:40AM |
0 |
ring noise at the background |
| 10:09AM |
0 |
Question: "When i Diall a group" |
| 10:02AM |
0 |
Ringduration problem when calling out via Sip |
| 9:45AM |
1 |
Buddy watch? |
| 9:44AM |
1 |
Bad Meetme() Bug |
| 9:44AM |
0 |
chan_zap.c:6570 handle_init_event error |
| 9:30AM |
1 |
Asterisk on MacOS? |
| 9:18AM |
7 |
NEWS: SIP Firmware Available for Cisco 7970 |
| 9:09AM |
0 |
Information to program a new driver for Asterisk |
| 8:54AM |
4 |
One Extension - Two Calls? |
| 8:33AM |
0 |
Set(LANGUAGE()=language) - for queue |
| 8:08AM |
4 |
spa3000 asterisk fxo gateway |
| 7:46AM |
1 |
grandstream handytone 286 sometimes dials out wrong number |
| 7:18AM |
1 |
Unable to make hints function properly |
| 6:28AM |
0 |
Outbound Proxy Support |
| 6:12AM |
0 |
hangup on silence? |
| 5:31AM |
1 |
Capturing DTMF during a call |
| 4:28AM |
1 |
Extension 's' in Realtime |
| 4:26AM |
0 |
problems in changing Festival's Default Voice in Asterisk |
| 3:48AM |
1 |
Unable to start Asterisk 1.2.5 with Asterisk-Addons 1.2.1 |
| 3:15AM |
3 |
Two asterisks on one machine |
| 1:21AM |
0 |
need to find an asterisk user from Costa Rica. |
| 1:15AM |
0 |
Passing Digits between ISDN PBX and Asterisk |
| 1:03AM |
0 |
No new mails |
| 12:42AM |
1 |
Redirecting to another service/server |
| |
| Sunday March 5 2006 |
| Time | Replies | Subject |
|---|
| 8:10PM |
0 |
static kernel |
| 7:37PM |
0 |
Dial() cmd executing Macro - dropped audio |
| 7:19PM |
1 |
Snom 360 Hinting tricks |
| 5:23PM |
6 |
Polycom 501 power over ethernet |
| 4:40PM |
0 |
ZapATA channels up, but calls cannot be made |
| 3:27PM |
0 |
Sipura SPA-3000 in Egypt |
| 1:20PM |
2 |
Problem with libpri? |
| 12:19PM |
0 |
RE: Asterisk-Users Digest, Vol 20, Issue 31 |
| 10:44AM |
0 |
re: Sixtel Services |
| 10:27AM |
0 |
to configure asterisk to work with the nathelper module of openser |
| 9:28AM |
0 |
re: Sixtel Services |
| 8:57AM |
2 |
Dialplan - strip IDD prefix and insert another |
| 8:57AM |
1 |
Inserting access codes as prefixes to CID |
| 6:10AM |
1 |
uniqueid |
| 6:06AM |
0 |
Realtime Content on LCD Display |
| 3:27AM |
1 |
20 seconds til voice transmission starts |
| 3:11AM |
1 |
Can log into the mailbox from Soft-phone , but not from Hardware Phone |
| |
| Saturday March 4 2006 |
| Time | Replies | Subject |
|---|
| 6:37PM |
2 |
Auto dial feature |
| 2:26PM |
2 |
Problem compiling ztdummy on centos 4, 2.6 kernel |
| 2:12PM |
2 |
Upgrading to 1.2.5? |
| 1:20PM |
1 |
# (send immediately) and dialplan broken on PAP2? |
| 9:08AM |
0 |
RE: Asterisk-Users Digest, Vol 20, Issue 20 |
| 8:55AM |
0 |
(no subject) |
| 8:54AM |
2 |
Asterisk 1.2.5 Released |
| 7:46AM |
1 |
Asterisk to a Huawei softX3000 |
| 7:29AM |
1 |
*** Yet another boring weekend? Test new Asterisk features in development! |
| 6:04AM |
2 |
help with asterisk installation |
| 2:37AM |
0 |
Accept Unregistered GK Calls |
| 2:35AM |
0 |
asterisk 1.2.5 cannot call a zap channel extension |
| 2:04AM |
1 |
What hardware to use for ISDN in Romania |
| 12:34AM |
0 |
Call Waiting? Should this just work? |
| |
| Friday March 3 2006 |
| Time | Replies | Subject |
|---|
| 10:21PM |
4 |
really need help with outgoing calls..PSTN errors |
| 9:54PM |
1 |
No audio on PRI. |
| 6:02PM |
10 |
MultiBRI in Australia - found one - maybe |
| 5:29PM |
2 |
Background() App From AGI |
| 5:18PM |
0 |
ANI configuration |
| 4:17PM |
0 |
Can we replace existing SIP call with new one? |
| 4:02PM |
2 |
Does an entry in AstDB stay after reboot? |
| 3:34PM |
0 |
Multi node call center |
| 3:29PM |
1 |
Meetme Timing Interface |
| 3:24PM |
2 |
Meetme Participant Announcement |
| 3:19PM |
1 |
Call Transfer - "Both legs must reside on Asterisk box to transfer at this time" |
| 3:16PM |
0 |
a=fmtp:18 annexb=no |
| 2:57PM |
1 |
Polycom 501 and single call only using AAH 2.2 |
| 2:32PM |
2 |
IAX to SIP conversion: SIP From header issue |
| 2:22PM |
5 |
new beta Grandstream firmware HT488_496_386 |
| 2:03PM |
0 |
Spontaneous reloads |
| 2:01PM |
0 |
CON-SNT-CP7970 resellers? |
| 1:06PM |
9 |
Preferred editor(s) dialplan coding? |
| 1:01PM |
0 |
Problem with HT-286 & BT-101 |
| 12:56PM |
3 |
Two PBX |
| 12:34PM |
0 |
[Fwd: Re: problem with incoming peer (cisco as5400)] |
| 12:14PM |
0 |
Bad quality between SIP and TDM |
| 12:10PM |
0 |
Asterisk coder conflicts |
| 11:46AM |
1 |
SIP Problem - Asterisk to Provider Gateway |
| 10:57AM |
1 |
IAX2 register problem |
| 10:49AM |
1 |
Hardware Requirements for 1M minutes |
| 10:48AM |
0 |
[HELP] dial plan continue for outbound channel on disconnect |
| 10:48AM |
1 |
dtmf tones problem with unicall and E1 |
| 9:26AM |
0 |
sprint FNTM(sp?) line |
| 9:19AM |
4 |
Echo Cancelation on TE110P |
| 9:17AM |
0 |
misdn <--> zap problem |
| 9:04AM |
2 |
what version s this?? |
| 9:03AM |
3 |
Sipura RMA |
| 8:37AM |
0 |
Fw: 2 real phone numbers on one SIP account |
| 8:32AM |
9 |
Program Buttons on Cisco 79xx Phones |
| 8:07AM |
0 |
Asterisk Realtime voicemail question |
| 8:06AM |
2 |
Autofill phonebook?? |
| 8:02AM |
0 |
Realtime Extensions hint priority |
| 7:36AM |
0 |
Thinking of moving from pure VoIP to PRI - thoughts? |
| 7:11AM |
0 |
'quit' isn't in the CLI's 'help' |
| 6:57AM |
0 |
check call status during call |
| 6:57AM |
0 |
calls only for logging users |
| 6:56AM |
2 |
Asterisk Fax Question |
| 6:31AM |
7 |
web meetme instructions |
| 6:09AM |
0 |
Implementing MOH while trunks gets connected... |
| 5:37AM |
1 |
login/logout agents in a specific queue |
| 3:53AM |
0 |
Part-Time work available |
| 3:13AM |
0 |
is there a variable for the calling IP ? |
| 2:45AM |
1 |
Problem with NAT!!! |
| 2:17AM |
0 |
Status of another channel from AGI |
| |
| Thursday March 2 2006 |
| Time | Replies | Subject |
|---|
| 11:11PM |
0 |
Australian E1 from Optus still not working |
| 11:10PM |
3 |
snom 320 MWI light |
| 7:32PM |
2 |
TIMESTAMP, DATETIME not working |
| 5:52PM |
1 |
RE: [on-asterisk] Brainstorming dual-core and Asterisk |
| 5:14PM |
0 |
RE: [on-asterisk] containers, virtualization, and high availability |
| 3:56PM |
2 |
Setting Max Calls on an IAX trunk |
| 3:40PM |
2 |
MixMonitor Problems -- sssshh, don't be too loud |
| 2:10PM |
1 |
Sip Realtime Configs Samples with MySQL |
| 1:34PM |
0 |
* dials out zap line first 6 digits, pause, then last digit |
| 1:19PM |
3 |
Child PID's |
| 1:17PM |
4 |
Changing caller id on transfer |
| 1:13PM |
0 |
Frequently Showed Info Messages |
| 12:17PM |
0 |
RE: Asterisk-Users Digest, Vol 20, Issue 13 |
| 12:14PM |
7 |
G729 and Meetme |
| 11:46AM |
0 |
OT - Cisco IP Phone and PC in different VLANs(with802.1x) |
| 11:44AM |
0 |
OT - Cisco IP Phone and PC in diferent VLANs(with 802.1x) |
| 11:24AM |
1 |
setmusiconhold doesn't work between 2 SIP phones |
| 10:46AM |
4 |
Polling Asterisk for Life |
| 10:42AM |
3 |
wake up calls |
| 10:27AM |
0 |
[HELP] Outbound Channel next priority on originator disconnect |
| 9:58AM |
1 |
Toshiba DK424 / Asterisk / DTMF problems |
| 9:51AM |
1 |
[Fwd: Over 40 destinations for FREE!] |
| 9:41AM |
0 |
problems with MOH |
| 9:16AM |
0 |
List disabled notification |
| 9:02AM |
0 |
problem with incoming peer (cisco as5400) |
| 8:51AM |
0 |
Native attended transfer: taking again original conversation |
| 8:46AM |
0 |
Redirect a sip outbound requests to a sip proxy |
| 8:02AM |
2 |
Get no busy signal on my analog line |
| 8:01AM |
0 |
channels appear to be stuck |
| 7:28AM |
0 |
Parked calls delay |
| 6:53AM |
0 |
gotoiftime with list of time range |
| 6:46AM |
2 |
test call quality |
| 6:43AM |
3 |
Sipura SPA-3000 vs Linksys SPA3000 |
| 6:23AM |
1 |
IAX Video and Meetme |
| 6:01AM |
0 |
remote IP address in channel? |
| 5:44AM |
1 |
error messages on /var/log/asterisk/messages |
| 4:30AM |
3 |
Native music on hold - Error |
| 3:54AM |
1 |
dial plan !! |
| 3:50AM |
5 |
Milliwatt Analyzer available |
| 3:01AM |
0 |
TE40X zapata.conf configuration sample |
| 2:46AM |
1 |
Managed Switches QoS to deal with network bottleneck |
| 1:55AM |
0 |
HDLC error |
| 12:59AM |
8 |
asterisk management interface |
| 12:45AM |
4 |
Info about F1000G |
| |
| Wednesday March 1 2006 |
| Time | Replies | Subject |
|---|
| 11:36PM |
0 |
hands-on experience with soft videophone |
| 7:37PM |
0 |
Want to record call and put into users voicemail |
| 6:56PM |
4 |
Polycom 501 |
| 5:41PM |
1 |
Settings for Yuxin Phones... |
| 5:32PM |
1 |
TE411P VPM |
| 4:16PM |
2 |
Two FXOs getting bridged? |
| 2:40PM |
0 |
queues & tranfers |
| 1:56PM |
2 |
GXP-2000 Volume Issue |
| 1:42PM |
3 |
my zap channel not ringing |
| 12:58PM |
0 |
Agents and Chanspy |
| 11:54AM |
0 |
perl AGI won't run from extensions.conf |
| 11:38AM |
2 |
OT - Cisco IP Phone and PC in diferent VLANs (with 802.1x) |
| 11:08AM |
0 |
Variables in queues.conf |
| 11:04AM |
2 |
ignore a DID? |
| 9:59AM |
3 |
160 analogue phones.. |
| 9:26AM |
3 |
Lowering Server Load |
| 8:51AM |
0 |
SendDTMF in connected call? |
| 7:52AM |
0 |
Configuration call hijack for users in a hunting group ? |
| 7:51AM |
9 |
Asterisk transfer conflict |
| 7:48AM |
1 |
SIP contexts being confused |
| 6:49AM |
0 |
Help with Digium TE210P, TDM400P card in Dell PE830 |
| 5:59AM |
0 |
Early media and custom SIP return codes |
| 5:39AM |
1 |
Agents, queues and Pentalties |
| 5:02AM |
6 |
Same CID on multiple users(friends9 in SIP.conf |
| 4:58AM |
9 |
MOH native files |
| 4:10AM |
1 |
how to run asterisk? |
| 3:04AM |
0 |
No sound after loading module wcte11xp |
| 3:03AM |
3 |
about operator |
| 2:39AM |
0 |
T38 fax pass thru to Cisco as53xx |
| 2:11AM |
0 |
Cisco 7905 - vad, cng |
| 1:49AM |
0 |
Callerid error: receiving DNID instead of callerID |
| 1:27AM |
0 |
ooh323 codec's - alaw |
| 1:17AM |
1 |
Cisco Callmanager integration with asterisk |
| 12:38AM |
2 |
Cannot log into mailbox , guidance requested |
| 12:13AM |
2 |
Working Asterisk with Austrian ISDN p2p |